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| 05:06.37 | eclark | when i run "asterisk -help", it spits out the version/usage info and then segfaults |
| 05:06.41 | eclark | i'm guessing this is not normal |
| 05:07.10 | eclark | i can't register the g.729 codec, because it looks like the register app runs "asterisk -help" |
| 05:09.06 | eclark | asterisk works, otherwise :-\ |
| 05:13.28 | eclark | sbin |
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| 13:04.04 | Entomologist | *** CLOSED (14411) [Applications/app_macro] calling Gosub in macro does not work if the invalid extension is triggered |
| 13:04.05 | Entomologist | Reported by: klaus3000 Karma: 21.5 |
| 13:04.05 | Entomologist | http://bugs.digium.com/view.php?id=14411 |
| 13:04.05 | Entomologist | ********************************************************* |
| 13:06.30 | *** join/#asterisk-bugs Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 13:06.30 | *** mode/#asterisk-bugs [+o Juggie] by ChanServ |
| 13:13.10 | leifmadsen | M14723 |
| 13:13.13 | MuffinMan | [acknowledged] [Asterisk] Core/Channels 0014723: ERROR[5003]: channel.c:2043 __ast_read: ast_read() called with no recorded file descriptor. reported by seadweller (Karma: neutral) http://bugs.digium.com/view.php?id=14723 |
| 13:13.21 | leifmadsen | M14732 |
| 13:13.23 | MuffinMan | [assigned] [Asterisk] Resources/General 0014732: Asterisk using 100% of cpu on 64 bit ubuntu reported by hoolahoous (Karma: neutral) http://bugs.digium.com/view.php?id=14732 |
| 13:13.35 | leifmadsen | putnopvut: do you think those issues are related? Someone seems to think they are, but I'm not sure. |
| 13:29.35 | leifmadsen | M14702 |
| 13:29.37 | MuffinMan | [new] [Asterisk] Core/PBX 0014702: [patch] debugging CID matching reported by klaus3000 (Karma: +21.50) http://bugs.digium.com/view.php?id=14702 |
| 13:29.44 | leifmadsen | Do we consider this a bug, or a new feature? |
| 13:29.48 | leifmadsen | is thinking feature.... |
| 13:30.44 | jsmith-away | leans toward bug as he runs out the door |
| 13:31.14 | leifmadsen | jsmith-away: okie, thanks for the feedback! |
| 13:34.20 | *** part/#asterisk-bugs T1000 (n=guaxinim@unaffiliated/guaxinim) |
| 13:56.55 | putnopvut | leifmadsen: the two issues you linked may be related. I need to look into 14732 a little more closes. |
| 13:57.00 | putnopvut | s/closes/closely/ |
| 13:59.07 | JunK-Y | putnopvut: about M14909 , do you know how can i break the inner and outter loop directly? |
| 13:59.11 | JunK-Y | M14909 |
| 13:59.12 | MuffinMan | [new] [Asterisk] Applications/app_chanspy 0014909: [patch] hangup when the spied-on channel hangups reported by junky (Karma: +160.25) http://bugs.digium.com/view.php?id=14909 |
| 14:02.11 | putnopvut | JunK-Y: you can't break out of two loops with just one statement in C, unless you use a goto. |
| 14:02.42 | JunK-Y | putnopvut: i remember old days, i did. |
| 14:02.54 | JunK-Y | something like a continue and a break or something like taht |
| 14:02.55 | JunK-Y | that |
| 14:16.25 | putnopvut | JunK-Y: have you tested the patch you placed on 14909? |
| 14:17.11 | JunK-Y | yes, it was working fine |
| 14:17.16 | JunK-Y | (with 1.4.20.1) |
| 14:17.36 | JunK-Y | the guy who posted the wiki is supposed to post feedback today |
| 14:17.38 | JunK-Y | why? |
| 14:20.04 | BlargMaN00 | file: ping |
| 14:20.19 | file | BlargMaN00: pong |
| 14:20.21 | putnopvut | JunK-Y: just curious. |
| 14:20.41 | BlargMaN00 | file: the patch you posted on 14850 doesn't seem to resolve the one way audio issue... |
| 14:20.58 | file | I didn't think so |
| 14:20.59 | JunK-Y | i will create a backport for some patches for 1.4 i think |
| 14:21.09 | JunK-Y | like the one you created for chanspy(,B) |
| 14:21.39 | BlargMaN00 | file: i also confirmed today that calls to the gateway going outside, work fine... it's just incoming from the gateway that are having issues... |
| 14:21.52 | file | did you upload the sip debug so we can see what may be different? |
| 14:22.17 | BlargMaN00 | file: i wasn't in the office yesterday, so I am going to do that this morning... |
| 14:22.24 | file | k |
| 14:30.34 | BlargMaN00 | file: uploaded debug of working outgoing call... |
| 14:31.31 | file | thanks! |
| 14:31.47 | BlargMaN00 | np |
| 14:36.33 | file | so CCM has two IPs - 10 and 11, you send outbound to 10? |
| 14:37.09 | BlargMaN00 | what i do, is check to see which one is available, and then send to the one that is available, and that is usually 10 |
| 14:37.14 | BlargMaN00 | sometimes 11 |
| 14:37.39 | file | there's two? |
| 14:37.51 | BlargMaN00 | yep... redundancy |
| 14:38.33 | file | that's odd redundancy, since the incoming calls from 11 actually have RTP going through 10 |
| 14:39.17 | BlargMaN00 | yeah... that's one of the things about CCM... they don't do anything the way you would think it should be done... |
| 14:39.44 | BlargMaN00 | if 10 goes down, then 11 does the rtp and sip handling |
| 14:40.06 | file | what about active calls if 10 goes down but the call was established through 11? |
| 14:40.46 | BlargMaN00 | prolly drop... never tested... I hate CCM, and wanna get rid of it anyway... but the IMAP VM isn't working well enough yet to put 200 users on it... 8) |
| 14:41.18 | BlargMaN00 | if i could set my server racks on fire to get rid of it i would, but that prolly wouldn't be a good thing |
| 14:41.45 | file | huh |
| 14:41.47 | BlargMaN00 | aster |
| 14:41.49 | BlargMaN00 | oops |
| 14:41.53 | file | outbound doesn't have session timers... |
| 14:42.18 | file | can you add something to your general section and test? |
| 14:42.47 | file | session-timers=refuse |
| 14:42.48 | BlargMaN00 | sure... but remember, i haven't changed anything from rc1 to rc4, and it worked in rc4 |
| 14:42.56 | BlargMaN00 | k hold |
| 14:43.25 | file | yeah, and the only change between rc2 and rc3 was for T38 |
| 14:43.41 | file | thus the general confusion over this |
| 14:43.47 | BlargMaN00 | hmm... * just core dumped on me... |
| 14:44.59 | file | stuff can't just work |
| 14:45.02 | file | always has to fight back. |
| 14:45.36 | file | can you reproduce the crash? |
| 14:46.06 | BlargMaN00 | no, I was trying to edit my sip.conf when it did it, and nothing was going on with system... no calls no nothing... |
| 14:46.35 | file | that wouldn't cause a crash... sip.conf is only touched on load or reload O.o |
| 14:47.19 | BlargMaN00 | i think it was just cause i've compiled and installed rc4 without a server reboot... it does that to me sometimes if i don't reboot |
| 14:47.47 | file | that's concerning. |
| 14:48.34 | BlargMaN00 | it does that to me with any of the 1.6.1.0 rc's |
| 14:48.45 | BlargMaN00 | but it's not all the time... it's sporatic... |
| 14:48.50 | file | can you get a backtrace from the core dump? |
| 14:49.15 | BlargMaN00 | i already rebooted, but i can try to reproduce it in a few minutes after i try this sip.conf change... |
| 14:49.29 | file | k |
| 14:50.14 | BlargMaN00 | still one-way audio |
| 14:50.24 | file | I'm at a loss. |
| 14:51.23 | BlargMaN00 | hmmm... |
| 14:51.25 | file | goes to track down the exact commits for chan_sip that are different between rc2 and rc3 |
| 14:51.55 | BlargMaN00 | i'll look through them too, and see if i can see something that looks suspicious... |
| 14:54.38 | file | 183067, 183110, 182042, 181987, 181849, 181298, 181359, 181283, 181035 |
| 14:57.40 | BlargMaN00 | file: what about 181768?? |
| 14:58.55 | BlargMaN00 | file: brb |
| 14:59.25 | file | 181771 let's see... |
| 15:01.32 | file | nah, don't think so |
| 15:02.02 | BlargMaN00 | just checking... didn't know if the 487 had anything to do with it... |
| 15:05.50 | file | I think I've narrowed it down. |
| 15:06.36 | file | 183110, 182042, 181035, 180263 |
| 15:06.44 | putnopvut | file: keep it up! |
| 15:06.46 | file | are the commits imo that would even remotely possibly touch it |
| 15:07.10 | *** join/#asterisk-bugs telecos (n=sergio@44.167.219.87.dynamic.jazztel.es) |
| 15:07.53 | file | BlargMaN00: do you have an SVN checkout of rc3? |
| 15:08.15 | BlargMaN00 | nope... i can get one though... |
| 15:08.39 | file | we can roll back the commits individually and test, if you have time? |
| 15:09.16 | BlargMaN00 | sure... i have some time... |
| 15:10.23 | BlargMaN00 | where do you want me to start?? at the rc3 release?? |
| 15:10.28 | file | yes |
| 15:10.32 | file | it makes it easier to track down |
| 15:10.43 | file | just do an svn co http://svn.digium.com/svn/asterisk/tags/1.6.1.0-rc3 |
| 15:12.08 | *** join/#asterisk-bugs leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
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| 15:18.35 | BlargMaN00 | ok... rc3 installed from svn and no workie... what next? |
| 15:19.56 | file | svn merge -c -183110 . |
| 15:20.01 | file | run that, it will revert revision 183110 |
| 15:20.06 | file | build, make, install, test |
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| 15:25.42 | BlargMaN00 | file: still not good |
| 15:25.58 | file | svn merge -c -182042 . |
| 15:26.00 | file | rinse and repeat |
| 15:26.37 | BlargMaN00 | ?? |
| 15:26.39 | BlargMaN00 | Select: (p) postpone, (df) diff-full, (e) edit, |
| 15:26.39 | BlargMaN00 | <PROTECTED> |
| 15:26.44 | BlargMaN00 | df right?? |
| 15:28.04 | file | ummm |
| 15:28.05 | file | p |
| 15:28.10 | file | then do svn status |
| 15:28.18 | file | and tell me the output |
| 15:29.02 | file | I suppose... I could just revert those all locally, provide a patch for you and if it is fixed we would know one of those 4 did it |
| 15:29.04 | file | I'll do that. |
| 15:29.20 | BlargMaN00 | http://pastebin.com/d14c865d8 |
| 15:29.21 | file | I wish I could reproduce this locally |
| 15:30.02 | BlargMaN00 | me too, although, i am developing some good debuging skills... 8)~ |
| 15:30.06 | file | oh that's fine |
| 15:30.08 | file | svn resolved . |
| 15:30.12 | file | make |
| 15:30.14 | file | make install |
| 15:30.19 | seanbright | test |
| 15:30.23 | seanbright | rinse and repeat |
| 15:30.27 | seanbright | is helping |
| 15:30.29 | seanbright | hreh |
| 15:31.54 | BlargMaN00 | file: not yet... |
| 15:32.07 | file | BlargMaN00: not yet? |
| 15:32.18 | BlargMaN00 | nope |
| 15:32.25 | file | oh, you mean that did not work? |
| 15:32.45 | BlargMaN00 | no... didn't work... next revision... |
| 15:32.53 | file | 181035 |
| 15:33.38 | BlargMaN00 | Conflict for property 'svn:externals' discovered on ''. |
| 15:33.45 | BlargMaN00 | i get that everytime i do the merge... |
| 15:33.51 | BlargMaN00 | but it keeps adding to it... |
| 15:34.05 | file | ignore >_> |
| 15:34.42 | BlargMaN00 | so, do i put diff-full, or postpone again?? |
| 15:34.49 | file | postpone |
| 15:34.51 | file | svn resolved . |
| 15:34.52 | file | make |
| 15:34.54 | file | make install |
| 15:35.04 | file | and I really wish the construction below me would cease and desist |
| 15:36.16 | Juggie | file, write a nasty letter |
| 15:36.30 | file | or not. |
| 15:36.46 | BlargMaN00 | file: no workie... |
| 15:36.51 | file | well gah |
| 15:38.24 | BlargMaN00 | 180263 no workie either... |
| 15:38.46 | file | there are 8 changes in chan_sip between rc2 and rc3 |
| 15:39.23 | BlargMaN00 | anything else before 180263?? |
| 15:39.34 | file | 183110, 182042, 181987, 181771, 181359, 181298, 181199, 181035 |
| 15:40.04 | file | 180263 was done in rc2 |
| 15:40.10 | file | I had to check and confirm. |
| 15:40.14 | mvanbaak | file: maybe a good idea to start with the chan_sip.c from rc2 in his rc3 env and start adding patches till it breaks ? |
| 15:40.26 | BlargMaN00 | oh... hmm... maybe it's not in chan_sip... maybe it's in the rtp.c |
| 15:40.51 | BlargMaN00 | just a thought |
| 15:41.17 | mvanbaak | BlargMaN00: that's easy to find. simply grab the chan_sip.c from rc2 and copy it over the channels/chan_sip.c you have right now |
| 15:41.20 | file | there was only one change between rc2 and rc3 for rtp.c, which had to do with packetization |
| 15:41.28 | file | yeah - do what mvanbaak says |
| 15:41.30 | file | mvanbaak: solid idea! |
| 15:42.05 | mvanbaak | gheh, from time to time I have an idea that's not a total waste of bandwidth :) |
| 15:42.20 | BlargMaN00 | ok... let me go grab another cup of coffee, and then i should be in good enough shape to do that... lol 8)~ |
| 15:42.27 | mvanbaak | whehehe |
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| 15:46.23 | file | I hope that works. |
| 15:46.28 | mvanbaak | me too |
| 15:49.43 | BlargMaN00 | ok... the rtp.c didn't work... but i also noticed that asterisk isn't sending the call canceled to the outside anymore... (i.e. my cell phone is not getting the signal that the call has terminated...) weird.... |
| 15:50.40 | file | okay - can you copy chan_sip.c from the channels directory from rc2 and put it into rc3? if it works we know for sure chan_sip.c is the issue |
| 15:51.56 | BlargMaN00 | http://pastebin.com/d62fd5c2 <-- you'll want to see this... happened after copying the chan_sip.c from rc2 into the svn... |
| 15:52.16 | BlargMaN00 | i've never seen that before... |
| 15:53.55 | putnopvut | ast_free undeclared? that can't be right. |
| 15:54.27 | BlargMaN00 | i didn't think so... that's why i said you'd prolly want to see that... |
| 15:54.28 | file | I believe if you open the file, jump to that line and replace ast_free with ast_free_ptr it should be happy |
| 15:54.41 | putnopvut | ah, forgot about that change. |
| 15:55.56 | BlargMaN00 | file: more gooder... you're such a fart smeller.............................. i mean smart feller... |
| 15:56.08 | file | I know nothing! |
| 15:56.13 | BlargMaN00 | lol |
| 15:57.13 | BlargMaN00 | ok... so we should now know, that the issue cannot possibly be related to chan_sip.c... cause that did not fix the issue... |
| 15:57.37 | file | that's concerning. |
| 16:02.00 | BlargMaN00 | still no go, even after a reboot... |
| 16:02.00 | file | I am back at a loss. |
| 16:02.09 | BlargMaN00 | i'd hate to think of how long it would take, but we could start at rc2 and apply 1 commit at a time until we found it... |
| 16:02.14 | BlargMaN00 | cringes at the thought... |
| 16:02.32 | putnopvut | BlargMaN00: it would be quicker to do a "bisect" |
| 16:02.34 | leifmadsen | how can we automate that? :) |
| 16:02.37 | putnopvut | Go to a halfway point |
| 16:02.42 | putnopvut | leifmadsen: git can ;) |
| 16:02.53 | leifmadsen | what can't git do? |
| 16:02.57 | BlargMaN00 | putnopvut: true that... |
| 16:03.13 | BlargMaN00 | git is the greatest |
| 16:05.01 | file | there is only one other thing outside of RTP and chan_sip that could be messing with the dialog |
| 16:05.02 | mvanbaak | leifmadsen: it cannot make coffee |
| 16:05.05 | file | CID |
| 16:05.58 | file | so weird. |
| 16:06.01 | BlargMaN00 | file: keep that going... i'll BRB 8) |
| 16:06.15 | leifmadsen | mvanbaak: or bring me wine |
| 16:06.23 | leifmadsen | goes off to get his teeth cleaned... weeee |
| 16:19.16 | BlargMaN00 | file: anything else you can think of to try? |
| 16:19.28 | file | no, not at the moment |
| 16:21.26 | BlargMaN00 | putnopvut: i think that IMAP is still having issues from rc1 to rc2... even after the patch, it still does some weird stuff, but works for the most part... |
| 16:22.05 | BlargMaN00 | putnopvut: i haven't pinned down exactly what it is doing yet, but it's not quite right... |
| 16:22.40 | putnopvut | BlargMaN00: are you subscribed to the asterisk-dev mailing list? There was something discussed there within the last day or two about some thread-specific storage problems. |
| 16:22.58 | putnopvut | It may possibly be that you are experiencing an effect of that, or it may be something else entirely. |
| 16:23.06 | BlargMaN00 | nope, but I can... |
| 16:23.25 | BlargMaN00 | putnopvut: i'll check it out... what is the thread subject?? |
| 16:24.27 | putnopvut | Just a sec... |
| 16:24.39 | putnopvut | "bug in app_voicemail.c" |
| 16:24.43 | putnopvut | I'll find a link to the thread. |
| 16:25.16 | putnopvut | http://lists.digium.com/pipermail/asterisk-dev/2009-April/037849.html |
| 16:25.31 | putnopvut | BlargMaN00: Oops, wrong one |
| 16:25.35 | putnopvut | Sorry :-/ |
| 16:25.50 | putnopvut | "pthread_getspecific() in app_voicemail" |
| 16:26.05 | putnopvut | http://lists.digium.com/pipermail/asterisk-dev/2009-April/037831.html |
| 16:26.12 | putnopvut | BlargMaN00: ^^ That's the right thread. |
| 16:27.41 | BlargMaN00 | putnopvut: i had another question for you, since we are on the topic of IMAP |
| 16:28.37 | putnopvut | Would you mind if I got back to you on it? It seems as though some of us are heading out to lunch pretty shortly. |
| 16:28.53 | putnopvut | Actually, you can ask it. I just may not be able to reply immediately. |
| 16:29.07 | BlargMaN00 | putnopvut: that's fine... it's kinda more as a new feature question than anything.. |
| 16:30.03 | BlargMaN00 | putnopvut: when IMAP gives this: [Apr 16 11:28:45] WARNING[6175]: app_voicemail.c:2309 mm_log: IMAP Warning: Unknown message data: 10 FETCH <-- is there anyway to have it call externnotify?? |
| 16:30.26 | BlargMaN00 | putnopvut: i think it would be more related to the c-client than anything... what do you think?? |
| 16:30.27 | putnopvut | Yeah, that would be possible. |
| 16:30.53 | putnopvut | I'd just have to make sure that we didn't try to make any c-client mail_XXXXX calls while doing that, because that would cause c-client to crash. |
| 16:31.21 | BlargMaN00 | putnopvut: i'm not sure how to implement it, cause I couldn't find where asterisk pastes that to the CLI, but that would keep me from running a cron script every second... |
| 16:31.48 | BlargMaN00 | putnopvut: and i'm sure that would help some other people out as well... |
| 16:32.04 | putnopvut | BlargMaN00: that's the mm_log() function in app_voicemail. when the c-client has information to log, it uses that callback. |
| 16:32.32 | BlargMaN00 | putnopvut: i'll check that out, and see if I can't write my first patch... 8)~ |
| 16:32.53 | putnopvut | All right, good luck :) |
| 16:33.27 | BlargMaN00 | thanks... i'm gonna need it... lol |
| 16:34.59 | BlargMaN00 | putnopvut: i assume that mm_notify is where the info comes from c-client?? |
| 16:49.16 | *** join/#asterisk-bugs BlackRat (n=ulmik@balticom-17-8.balticom.lv) |
| 16:49.56 | BlackRat | most probably it aint a bug but misconfiguration .. any clues what means: |
| 16:49.57 | BlackRat | [Apr 16 19:42:34] WARNING[5636]: channel.c:2948 set_format: Unable to find a codec translation path from 0x400 (ilbc) to 0x8 (alaw) |
| 16:49.57 | BlackRat | [Apr 16 19:42:34] WARNING[5481]: chan_dahdi.c:5406 dahdi_write: Cannot handle frames in 1024 format |
| 16:51.39 | file | it couldn't transcode from ilbc to alaw. |
| 16:53.36 | file | instructions are present in UPGRADE.txt for getting codec_ilbc to build |
| 16:54.38 | BlackRat | file: thnx but whats about that dahdi error? |
| 16:54.59 | BlackRat | or they are related ? |
| 16:55.04 | file | it sent the ilbc audio into chan_dahdi |
| 16:55.10 | file | and chan_dahdi does not understand ilbc |
| 17:22.58 | BlackRat | file: thnx it solved an issue |
| 18:30.21 | *** join/#asterisk-bugs Deeewayne (n=dwayne@95.104.105.2) |
| 18:30.21 | *** mode/#asterisk-bugs [+o Deeewayne] by ChanServ |
| 18:31.27 | BlargMaN00 | file: ping |
| 19:01.26 | file | BlargMaN00: pong |
| 19:02.57 | BlargMaN00 | file: http://pastebin.com/d112367d4 <-- tried using a fresh checkout of the rc3 and then copying channel.c from rc2 and got this when i compiled... |
| 19:05.16 | file | yes, it was added to channel.c afterwards |
| 19:06.32 | BlargMaN00 | gotcha... |
| 19:08.59 | BlargMaN00 | ls |
| 19:09.01 | BlargMaN00 | oops |
| 19:28.39 | file | BlargMaN00: so you're just taking files from rc2 and dropping them into rc3 to narrow down the issue? brute force. |
| 19:29.23 | BlargMaN00 | basically... but can't really do anything with channel.c cause of the new feature... so I am going to start with rc2 and work my way to rc3 until i find where it stops working... |
| 19:30.09 | BlargMaN00 | what would be the best way to merge the svn forward including all revisions up to a certain point? |
| 19:30.21 | file | I don't... know... |
| 19:30.30 | file | you could have just dropped in features.c as well |
| 19:31.43 | BlargMaN00 | tried that to... got this --> http://pastebin.com/d276d8062 |
| 19:32.00 | file | ast_free_ptr |
| 19:32.09 | BlargMaN00 | but what i didn't do after that, was try to .................. just about to say that... |
| 19:32.18 | file | I'm really bothered that this isn't chan_sip |
| 19:32.26 | file | because what is happening is strictly chan_sip |
| 19:32.41 | BlargMaN00 | or....... so it should be..... 8/ |
| 19:37.05 | file | what is your network topology like? |
| 19:37.34 | BlargMaN00 | as far as voice or data or both? |
| 19:38.18 | file | CCM stuff, Asterisk, and phones all can talk to eachother directly - no firewalls/NAT in between? |
| 19:39.22 | BlargMaN00 | ccm and phones are all on the same vlan... asterisk is on a different vlan, but that's just cause i never put it on the same vlan... wouldn't be hard, but the all talk well together through a router... no nat tho |
| 19:41.54 | BlargMaN00 | never had any issues until this |
| 19:45.48 | file | huh |
| 19:45.50 | file | svn-bisect |
| 19:53.23 | BlargMaN00 | file: ok... so a fresh checkout of rc3 with the channel.c and features.c (with the ast_free_ptr change) from rc2... we are good to go... |
| 19:53.34 | BlargMaN00 | now we just need to figure out which commit broke the issue... |
| 19:53.40 | file | it works? |
| 19:53.42 | file | oh dear. |
| 19:53.46 | file | I have a thougt. |
| 19:53.48 | BlargMaN00 | yep |
| 19:53.49 | file | and thought |
| 19:54.01 | BlargMaN00 | aight... shoot |
| 19:54.38 | Corydon76-dig | It's already gone |
| 19:55.04 | file | uno momento |
| 19:56.07 | file | 182810, 182570, 182533, 182527 |
| 19:56.43 | file | specifically 182527 |
| 19:56.50 | seanbright | commit asterisk 182527 |
| 19:56.55 | MuffinMan | [182527] Committed by kpfleming on 2009-03-17T14:39:50.446476Z: Merged revisions 182525 via svnmerge from |
| 19:56.55 | MuffinMan | ..https://origsvn.digium.com/svn/asterisk/trunk |
| 19:56.56 | MuffinMan | .. |
| 19:56.56 | MuffinMan | .......... |
| 19:56.56 | MuffinMan | .. r182525 | kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11 lines |
| 19:56.59 | MuffinMan | .. |
| 19:57.01 | MuffinMan | .. Improve behavior of ast_answer() to not lose incoming frames |
| 19:57.04 | MuffinMan | .. |
| 19:57.06 | MuffinMan | .. ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations. |
| 19:57.10 | MuffinMan | .. |
| 19:57.12 | MuffinMan | .. When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames. |
| 19:57.13 | *** mode/#asterisk-bugs [+b MuffinMan!*@*] by seanbright |
| 19:57.16 | seanbright | yeah, we get it. |
| 19:57.34 | file | BlargMaN00: can you roll back those in order and give it a go? |
| 19:57.39 | file | BlargMaN00: btw you rock |
| 19:58.52 | BlargMaN00 | file: np... that's what i'm here for... |
| 19:58.56 | *** mode/#asterisk-bugs [-b MuffinMan!*@*] by seanbright |
| 19:59.22 | file | seanbright: he really doesn't like it when you do that |
| 19:59.33 | seanbright | ban/unban? |
| 19:59.38 | file | seanbright: paying for therapy for MuffinMan comes out of my pocket! |
| 19:59.48 | seanbright | well there is no other way to shut him up |
| 19:59.56 | seanbright | other than not asking him anything in the first place |
| 19:59.59 | seanbright | which was totally my bad. |
| 20:00.13 | file | you *could* have asked him in privmsg |
| 20:00.19 | seanbright | you are correct |
| 20:00.23 | seanbright | my apologies to both of you |
| 20:00.36 | file | I'll send the bill for his therapy to you |
| 20:00.44 | seanbright | wonderful |
| 20:01.06 | seanbright | 123 KissMyAss Street |
| 20:01.12 | seanbright | Baltimore, MD 21236 |
| 20:01.15 | seanbright | :) |
| 20:03.08 | BlargMaN00 | file: ok... so when i do a svn revert . i don't get the rc3 files back... is that an issue?? |
| 20:03.09 | file | svn revert -R . |
| 20:03.09 | BlargMaN00 | ahhh... that's sooooo much better... so, how about that therapy... 8)~ |
| 20:06.41 | BlargMaN00 | file: just FYI... that also fixed the problem where my cell phone was seeing the call terminate... so ya know... |
| 20:08.10 | *** part/#asterisk-bugs BlackRat (n=ulmik@balticom-17-8.balticom.lv) |
| 20:08.40 | *** join/#asterisk-bugs leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:08.40 | *** mode/#asterisk-bugs [+o leifmadsen] by ChanServ |
| 20:13.13 | file | we used to wait for audio before trying to native bridge |
| 20:13.18 | file | starting in rc3 we don't do that |
| 20:13.27 | file | if that is what is causing this problem then CCM is dumb |
| 20:14.04 | BlargMaN00 | file: wow... when i rolled back 182527, it pretty much needed everything in * to recompile... but.... it fixed the issue.... |
| 20:14.21 | file | that is stupid. |
| 20:14.29 | BlargMaN00 | that's ccm for ya |
| 20:14.55 | file | I can't express how stupid that is. |
| 20:15.05 | BlargMaN00 | but at the same time though... cisco could have changed that by now, as they have released ver 8.something, and we are still on 4.2 cause it costs too much to upgrade... |
| 20:15.30 | file | I would bet this is a bug in their reinvite handling |
| 20:15.42 | mvanbaak | CCM has no bugs |
| 20:15.44 | mvanbaak | ;) |
| 20:16.00 | BlargMaN00 | ccm is "perfect"... |
| 20:16.07 | mvanbaak | indeed |
| 20:16.20 | file | we tell the Polycom to send to CCM so we don't send it anything |
| 20:16.34 | file | we then send a reinvite to tell CCM to send audio to the Polycom |
| 20:16.45 | file | the CCM maybe blocks until it receives RTP from us? |
| 20:17.14 | file | which will never happen |
| 20:17.34 | BlargMaN00 | maybe so... |
| 20:17.51 | file | that is stupid. |
| 20:18.19 | BlargMaN00 | i wish i had a newer version to test with and find out if it's the version causing the problem, but i don't... |
| 20:18.45 | file | okay, I'm fairly certain we've exposed a bug in the version of CCM you are using |
| 20:18.52 | BlargMaN00 | and i won't cause it cost way too much, not to mention the fact that cisco says "you have to upgrade your hardware, cause it won't run correctly..." bah humbug... |
| 20:19.11 | file | but it's only a one line change to make Asterisk wait for audio before bridging |
| 20:19.16 | file | so let's try that to confirm |
| 20:19.20 | file | svn revert -R . |
| 20:19.48 | BlargMaN00 | file: oooo oooo ooo... i just got that core dump i was telling you about... let me get the backtrace real quick.... |
| 20:19.55 | file | neat! pastebin it please |
| 20:22.29 | BlargMaN00 | file: http://pastebin.com/d70fcff14 |
| 20:22.37 | BlargMaN00 | file: no bt info though.... weird... |
| 20:23.02 | file | how did you open it? |
| 20:24.21 | BlargMaN00 | file: http://pastebin.com/d7ffb475a <-- highlighted with command... |
| 20:24.41 | file | should be okay... ungood |
| 20:25.22 | BlargMaN00 | looks to me like the kernel entered some sort of endless loop that killed, but that's just my best _guess_ |
| 20:25.53 | file | hrm, let's continue on with the no audio issue |
| 20:26.02 | BlargMaN00 | ok... svn revert -R . what next... |
| 20:26.15 | file | open main/features.c and find the one usage of ast_raw_answer |
| 20:26.21 | file | replace it with ast_answer(chan) |
| 20:27.23 | BlargMaN00 | done... compiling... |
| 20:30.17 | BlargMaN00 | file: issue resolved... |
| 20:30.29 | file | that is stupid. |
| 20:31.03 | BlargMaN00 | yeah pretty much... i wouldn't worry about it too much... i'm prolly the only person in the world having that issue... |
| 20:32.04 | file | that change should be applicable for all versions above 1.4 and for the future |
| 20:32.48 | file | and should be safe for your environment |
| 20:32.49 | BlargMaN00 | i'm sure it is acceptable, and that's why i made mention, that I am using such an old version of CCM... i'm sure it has been resolved on the newer versions..... hopefully... |
| 20:32.56 | file | nods |
| 20:33.03 | file | awesome |
| 20:33.15 | file | thanks for narrowing things down! |
| 20:34.01 | BlargMaN00 | no worries... i love developing software, i just don't understand the structure of the * code yet, or i'd help more... |
| 20:34.10 | file | you now have 3.50 karma! |
| 20:34.18 | BlargMaN00 | sweet.... |
| 20:35.51 | file | resolved and closed |
| 20:36.50 | BlargMaN00 | cool... i'm going to call my cisco vendor, and have them see if cisco has anything that will resolve that for my version... that would be awesome if they did... |
| 20:37.03 | file | I would not hold your breath. |
| 20:37.28 | BlargMaN00 | i'm not... i'd like to live for a lot more years... 8/ |
| 20:41.22 | BlargMaN00 | file: how hard would it be to add a patch that allows you to turn that on and off?? |
| 20:41.47 | BlargMaN00 | like 'waitforrtp=yes' |
| 20:42.25 | file | ummm not hard |
| 20:43.52 | *** join/#asterisk-bugs putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 20:43.52 | *** mode/#asterisk-bugs [+o putnopvut] by ChanServ |
| 20:43.55 | *** join/#asterisk-bugs jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 20:44.27 | *** join/#asterisk-bugs mnicholson (n=mnichols@nat/digium/x-6e0db4074b36f880) |
| 20:44.28 | *** join/#asterisk-bugs The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel) |
| 20:47.00 | BlargMaN00 | file: where would i look to see where variables are read and parsed from sip.conf?? |
| 20:47.10 | file | well this isn't being done at the SIP level |
| 20:47.28 | file | you could make a global option for asterisk.conf that enables or disables waiting for audio before bridging |
| 20:48.00 | BlargMaN00 | yeah... i could try that... where is that parsed in the code?? |
| 20:48.14 | BlargMaN00 | i have to start small and work up to bigger projects... 8) |
| 20:48.20 | file | asterisk.c |
| 20:48.39 | BlargMaN00 | gotcha... i'll (hopefully) figure it out from there... |
| 21:02.31 | Entomologist | *** CLOSED (14912) [Applications/app_voicemail] voicemail umask / permissions bug |
| 21:02.31 | Entomologist | Assigned to: tilghman |
| 21:02.31 | Entomologist | Reported by: jcapp Karma: 4 |
| 21:02.31 | Entomologist | http://bugs.digium.com/view.php?id=14912 |
| 21:02.31 | Entomologist | ********************************************************* |
| 21:13.51 | file | BlargMaN00: Qwell pointed out that you ran gdb on safe_asterisk instead of asterisk when getting that bt |
| 21:34.41 | Entomologist | *** CLOSED (14916) [Channels/chan_sip/Interoperability] SIP video phone requires numberofmediastreams > 4 (chan_sip.c) |
| 21:34.41 | Entomologist | Reported by: moshili Karma: 0 |
| 21:34.41 | Entomologist | http://bugs.digium.com/view.php?id=14916 |
| 21:34.41 | Entomologist | ********************************************************* |
| 21:36.22 | leifmadsen | file: ... |
| 21:36.26 | leifmadsen | M14913 |
| 21:36.28 | MuffinMan | [new] [Asterisk] Channels/chan_sip/General 0014913: limitonpeers=yes vs call-limit=1 didn't work on asterisk 1.4.24 reported by sybasesql (Karma: +1.50) http://bugs.digium.com/view.php?id=14913 |
| 21:36.39 | leifmadsen | do you think that is just because the incoming call matched on the 'peer' structure? (i.e. IP based matching?) |
| 21:40.35 | file | uh yes, it looks like it is doing exactly as configured |
| 21:40.57 | Qwell | yay! |
| 21:41.16 | Entomologist | *** CLOSED (14885) [Channels/chan_sip/Registration] [patch] rtupdate=no not working |
| 21:41.16 | Entomologist | Assigned to: tilghman |
| 21:41.16 | Entomologist | Reported by: deepesh Karma: 0 |
| 21:41.16 | Entomologist | http://bugs.digium.com/view.php?id=14885 |
| 21:41.16 | Entomologist | ********************************************************* |
| 21:41.59 | leifmadsen | file: close? |
| 21:42.39 | file | yes |
| 21:42.52 | leifmadsen | thanks! |
| 21:43.50 | Entomologist | *** CLOSED (14913) [Channels/chan_sip/General] limitonpeers=yes vs call-limit=1 didn't work on asterisk 1.4.24 |
| 21:43.50 | Entomologist | Reported by: sybasesql Karma: 1.5 |
| 21:43.50 | Entomologist | http://bugs.digium.com/view.php?id=14913 |
| 21:43.50 | Entomologist | ********************************************************* |
| 21:44.26 | leifmadsen | M14917 |
| 21:44.28 | MuffinMan | [new] [Asterisk] Channels/chan_sip/Interoperability 0014917: SIP video phone requires numberofmediastreams > 4 (chan_sip.c) reported by moshili (Karma: neutral) http://bugs.digium.com/view.php?id=14917 |
| 21:44.32 | leifmadsen | Is that a sane change? |
| 21:46.33 | *** join/#asterisk-bugs popolinux (n=cyril@AToulouse-156-1-114-165.w90-30.abo.wanadoo.fr) |
| 21:48.55 | Corydon76-dig | leifmadsen: replied, asking for more info |
| 21:49.27 | leifmadsen | Corydon76-dig: merci! |
| 21:49.44 | leifmadsen | Corydon76-dig: do you remember what the thread name was for the 'i' extension discussion recently with klaus3000? |
| 21:49.55 | leifmadsen | I think I can close another issue related to the 'i' extension |
| 21:50.02 | leifmadsen | M14907 |
| 21:50.03 | MuffinMan | [assigned] [Asterisk] Functions/func_dialplan 0014907: [1.6.0.9] exten => i ignored reported by kowalma (Karma: +2.00) http://bugs.digium.com/view.php?id=14907 |
| 21:50.13 | Corydon76-dig | It's probably sane, but if we make the change, I'd like to do it knowing what the media streams are |
| 21:52.21 | Corydon76-dig | I agree, 14907 can be closed |
| 21:52.49 | leifmadsen | thanks! I'd like to reference the discussion if possible (if you remember the subject text) |
| 21:53.01 | Corydon76-dig | Don't remember, sorry |
| 21:53.07 | leifmadsen | np, I'll just close |
| 21:56.24 | Entomologist | *** CLOSED (14907) [Functions/func_dialplan] [1.6.0.9] exten => i ignored |
| 21:56.24 | Entomologist | Assigned to: lmadsen |
| 21:56.24 | Entomologist | Reported by: kowalma Karma: 2 |
| 21:56.24 | Entomologist | http://bugs.digium.com/view.php?id=14907 |
| 21:56.24 | Entomologist | ********************************************************* |
| 22:08.44 | *** part/#asterisk-bugs popolinux (n=cyril@AToulouse-156-1-114-165.w90-30.abo.wanadoo.fr) |
| 22:24.34 | leifmadsen | M14913 |
| 22:24.35 | MuffinMan | [new] [Asterisk] Channels/chan_sip/General 0014913: limitonpeers=yes vs call-limit=1 didn't work on asterisk 1.4.24 reported by sybasesql (Karma: +1.50) http://bugs.digium.com/view.php?id=14913 |
| 22:24.46 | leifmadsen | file: can you see if I said something wrong there and re-close if appropriate? |
| 23:32.54 | *** join/#asterisk-bugs eliel (n=eliels@151-202-114-200.fibertel.com.ar) |